MTG3000
High Density Digital VoIP Gateway
High Density Digital VoIP Gateway
MTG3000 is a carrier-grade Digital VoIP gateway, scalable from 16 to 63 ports E1/T1 with STM-1 interface. It provides carrier-grade VoIP and FoIP services, as well as value-added functions such as modem and voice recognition. With highly maintainable, manageable and operable features, it offers a high performance, reliable communication network for users.
MTG3000 supports a wide-range of signaling protocols, realizing the interconnection between SIP and traditional signals like ISDN PRI / SS7, utilizing efficiency of trunking resources while ensuring voice quality. With multiple voice codes, secure signal encryption and smart voice recognition technology, MTG3000 is ideal for a variety of applications of services providers and telecom operators.
MTG3000 supports a wide-range of signaling protocols, realizing the interconnection between SIP and traditional signals like ISDN PRI / SS7, utilizing efficiency of trunking resources while ensuring voice quality. With multiple voice codes, secure signal encryption and smart voice recognition technology, MTG3000 is ideal for a variety of applications of services providers and telecom operators.
Detailed parameters

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1+1 Redundant Main Control Unit (MCU)
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4 Digital Processing Unit (DTU), each support 512 channels
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Dual Power Supplies
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2 GE
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SIP v2.0
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SIP-T,RFC3372, RFC3204, RFC3398
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SIP Trunk Work Mode: Peer/Access
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SIP/IMS Registration :with up to 256 SIP Accounts
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NAT: Dynamic NAT, Rport
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Flexible Route Methods: PSTN-PSTN, PSTN-IP, IP-PSTN
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Intelligent Routing Rules
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Call Routing base on Time
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Call Routing base on Caller/Called Prefixes
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256 Route Rules for each Direction
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Caller and Called Number Manipulation
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Local/Transparent Ring Back Tone
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Overlapping Dialing
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Dialing Rules, with up to 2000
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PSTN group by E1 port or E1 Timeslot
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IP Trunk Group Configuration
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Voice Codecs Group
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Caller and Called Number White Lists
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Caller and Called Number Black Lists
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IP Trunk Priority
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Radius
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Up to 63 E1s/T1s, STM-1 interface
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Codecs:G.711a/μ law,G.723.1, G.729A/B, iLBC 13k/15k,AMR
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Silence Suppression
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Comfort Noise
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Voice Activity Detection
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Echo Cancellation (G.168),with up to 128ms
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Adaptive Dynamic Buffer
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Voice, Fax Gain Control
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FAX:T.38 and Pass-through
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DTMF Mode: RFC2833/SIP Info/In-band
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Clear Channel/Clear Mode
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ISDN PRI:
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Signal 7/SS7: ITU-T, ANSI, ITU-CHINA, MTP1/MTP2/MTP3, TUP/ISUP
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R2 MFC
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Web GUI Configuration
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Data Backup/Restore
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PSTN Call Statistics
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SIP Trunk Call Statistics
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Firmware Upgrade via TFTP/Web
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SNMP v1/v2/v3
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Network Capture
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Syslog: Debug, Info, Error, Warning , Notice
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Call History Records via Syslog
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NTP Synchronization
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Centralized Management System